[Freeswitch-users] Grandstream gateways
Milena
testeador01 at gmail.com
Wed Nov 25 13:00:14 PST 2009
Hello,
Samuel: We also have some GXW4104 gateways, in small production/testing
environments; our caller id works fine and none of them has failed in over a
year of being used. The thing that i dislike about the GXW series is that it
has no support for early media.
Everyone: What FXO devices do you currently use / recommend?
2009/11/25 Chris Chen <chris.chen2004 at gmail.com>
> You haven't really put it into production for more than one year. The issue
> with GXW4108 is that after some time, say a couple of months, either all FXO
> ports not working, or worse, some FXO ports not working, but after power
> recycling, they will come back to work for some time until on strike again
> at some time you have no control.
>
> This had been reported for a couple of years without improvement. Go google
> search you will find out, this has happened to many GXW4108 users.
>
>
>
> On Wed, Nov 25, 2009 at 3:16 PM, Samuel Mukoti <samuelmukoti at gmail.com>wrote:
>
>> Thank you for those tips,
>>
>> I do have some small setups using gxw4108 they work or, except CID
>> doesn't seem to work. I will try the channel bank route - just don't
>> know too much about the setup options or how you'd purchase the
>> correct config, eg. For 150 FXS channel bank, can I get a single PCI
>> card for that?
>>
>> I may end up using the grandstream fxs gateways then use the T1
>> channel bank from sangoma,
>>
>> Thank you all..
>>
>> Lastly, I know asterisk now has an offical skype_ module, Is there
>> anything similar I could use?
>>
>>
>> On 25 Nov,2009, at 9:52 PM, Cory Andrews <cory at voipsupply.com> wrote:
>>
>> > Samuel - you could go with FXS gateways or channel banks. If you go
>> > the gateway route Grandstream or Audiocodes would work fine.
>> > Audiocodes are a bit more telco grade. If you have 25 POTS incoming
>> > you could use a 24FXO channel bank cross connected with Rhino T1
>> > cards, or individual FXO gateways but you may have a hard time
>> > finding 24 ports of FXO in a single GW. Best performing T1 cards in
>> > my experience (thousands of deployments) are Sangoma. Your server
>> > configuration looks fine.
>> >
>> > Cory J. Andrews
>> > Director New Market Initiatives
>> >
>> > Sayers Media Group
>> > VoIP Supply, LLC
>> > 454 Sonwil Drive
>> > Buffalo, NY 14225
>> > 716-250-3402 OFFICE
>> > 716-630-1548 FAX
>> > 716-601-4474 MOBILE
>> > candrews at sayersmedia.com
>> >
>> >
>> > Have I exceeded your expectations? Please share your experience
>> > with my boss, Benjamin P. Sayers, CEO
>> >
>> > NOTICE: The information contained in this email and any document
>> > attached hereto is intended only for the named recipient(s). It is
>> > the property of the VoIP Supply, LLC and shall not be used,
>> > disclosed or reproduced without the express written consent of VoIP
>> > Supply, LLC. If you are not the intended recipient, nor the employee
>> > or agent responsible for delivering this message in confidence to
>> > the intended recipient(s), you are hereby notified that you have
>> > received this transmittal in error, and any review, dissemination,
>> > distribution or copying of this transmittal or its attachments is
>> > strictly prohibited. If you have received this transmittal and/or
>> > attachments in error, please notify me immediately by reply e-mail
>> > or telephone and then delete this message, including any
>> > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY
>> > 14225 USA.
>>
>> >
>> >
>> >
>> > -----Original Message-----
>> > From: freeswitch-users-bounces at lists.freeswitch.org
>> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> > Samuel Mukoti
>> > Sent: Wednesday, November 25, 2009 2:40 PM
>> > To: freeswitch-users at lists.freeswitch.org
>> > Subject: [Freeswitch-users] Grandstream gateways
>> >
>> > Hi all,
>> >
>> > I'm wanting to try out a my first large scale setup at the office, 200
>> > extensions and 24 POTS incoming, also a T1 line once the telco guys
>> > are ready. I wanted assistance with choosing the most appropriate
>> > hardware. We already have about 150 analogue phones, and I was
>> > wondering what's best? A couple of grandstream FXS GXW4024? Also for
>> > my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma
>> > or digium card? The best voice quality is paramount. Lastly for T1
>> > what cards are recommeded,
>> >
>> > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM,
>> > would that perform? Or do I need hardware transcoding?
>> >
>> > Thank you,
>> >
>> > Sam
>> >
>> > Twitter: twitter.com/samuelmukoti
>> >
>> >
>> > On 25 Nov,2009, at 8:05 PM,
>> freeswitch-users-request at lists.freeswitch.org
>> > wrote:
>> >
>> >> Send FreeSWITCH-users mailing list submissions to
>> >> freeswitch-users at lists.freeswitch.org
>> >>
>> >> To subscribe or unsubscribe via the World Wide Web, visit
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> or, via email, send a message with subject or body 'help' to
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>> >>
>> >> You can reach the person managing the list at
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>> >>
>> >> When replying, please edit your Subject line so it is more specific
>> >> than "Re: Contents of FreeSWITCH-users digest..."
>> >>
>> >>
>> >> Today's Topics:
>> >>
>> >> 1. Re: mod_conference kick to abort invitations (Michael Jerris)
>> >> 2. Re: Handling the 302 Moved Temporarily response from
>> >> JavaScript (Michael Jerris)
>> >> 3. Re: No NOTIFY MWI when registering via proxy. (Brian West)
>> >> 4. Re: remote_media_ip variable not set (Michael Jerris)
>> >> 5. Re: How to find whether the destination extension supports
>> >> encryption (Michael Jerris)
>> >> 6. Re: Bypass_media and re_invite (srinivasula reddy)
>> >> 7. Re: Handling the 302 Moved Temporarily response from
>> >> JavaScript (Stephen Crosby)
>> >> 8. Re: Handling the 302 Moved Temporarily response from
>> >> JavaScript (Tihomir Culjaga)
>> >>
>> >>
>> >> ---
>> >> -------------------------------------------------------------------
>> >>
>> >> Message: 1
>> >> Date: Wed, 25 Nov 2009 12:44:46 -0500
>> >> From: Michael Jerris <mike at jerris.com>
>> >> Subject: Re: [Freeswitch-users] mod_conference kick to abort
>> >> invitations
>> >> To: freeswitch-users at lists.freeswitch.org
>> >> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com>
>> >> Content-Type: text/plain; charset="windows-1252"
>> >>
>> >> Its a feature we don't have, patches welcome.
>> >>
>> >> Mike
>> >>
>> >> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote:
>> >>
>> >>> Hi members,
>> >>> I?m controlling freeswitch with the conference module via xmlrpc.
>> >>>
>> >>> Is it desired that the kick command can only kick users that are
>> >>> connected to the conference?
>> >>> Is there no chance abort an invitation?
>> >>> The kick command has no effect until the person I invited with the
>> >>> dial command is connected.
>> >>
>> >> -------------- next part --------------
>> >> An HTML attachment was scrubbed...
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>> >>
>> >> ------------------------------
>> >>
>> >> Message: 2
>> >> Date: Wed, 25 Nov 2009 12:45:50 -0500
>> >> From: Michael Jerris <mike at jerris.com>
>> >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily
>> >> response from JavaScript
>> >> To: freeswitch-users at lists.freeswitch.org
>> >> Message-ID: <A8FA625F-16D2-4A9F-B8C4-13343A488777 at jerris.com>
>> >> Content-Type: text/plain; charset=us-ascii
>> >>
>> >> In trunk there is a sofia profile setting to allow dialplan
>> >> processing of 302 responses. This won't get you back into your same
>> >> javascript, but you can probably do something clever from there.
>> >>
>> >> Mike
>> >>
>> >> On Nov 24, 2009, at 5:04 PM, John Platts wrote:
>> >>
>> >>>
>> >>> I have considered writing JavaScript code to bridge two calls
>> >>> together. However, I would like to perform custom handling of the
>> >>> 302 Moved Temporarily response. How do I handle the 302 Moved
>> >>> Temporarily response if I use JavaScript?
>> >>>
>> >>
>> >>
>> >>
>> >> ------------------------------
>> >>
>> >> Message: 3
>> >> Date: Wed, 25 Nov 2009 11:46:05 -0600
>> >> From: Brian West <brian at freeswitch.org>
>> >> Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via
>> >> proxy.
>> >> To: freeswitch-users at lists.freeswitch.org
>> >> Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org>
>> >> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes
>> >>
>> >> Yes an alias will be required for every domain you run on the profile
>> >> so it can find it.
>> >>
>> >> /b
>> >>
>> >> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote:
>> >>
>> >>> Try an alias on the sip profile.
>> >>>
>> >>> Mike
>> >>
>> >>
>> >>
>> >>
>> >> ------------------------------
>> >>
>> >> Message: 4
>> >> Date: Wed, 25 Nov 2009 12:47:37 -0500
>> >> From: Michael Jerris <mike at jerris.com>
>> >> Subject: Re: [Freeswitch-users] remote_media_ip variable not set
>> >> To: freeswitch-users at lists.freeswitch.org
>> >> Message-ID: <DF3ECA04-0247-40BB-A810-2468F9C4D805 at jerris.com>
>> >> Content-Type: text/plain; charset=us-ascii
>> >>
>> >> It's possible it does not. I just added some code to set it on auto-
>> >> adjust so it might be there sometimes now. You might need to add
>> >> some code in mod_sofia to add it other times. Maybe it makes sense
>> >> to move that var setting down to switch_rtp.c. Patches for this
>> >> would be welcome.
>> >>
>> >> Thanks
>> >>
>> >> Mike
>> >>
>> >> On Nov 24, 2009, at 10:56 AM, Juan Backson wrote:
>> >>
>> >>> Hi,
>> >>>
>> >>> In the case of proxy_media=true, does it gets set at all then?
>> >>
>> >>
>> >>
>> >>
>> >> ------------------------------
>> >>
>> >> Message: 5
>> >> Date: Wed, 25 Nov 2009 12:48:39 -0500
>> >> From: Michael Jerris <mike at jerris.com>
>> >> Subject: Re: [Freeswitch-users] How to find whether the destination
>> >> extension supports encryption
>> >> To: freeswitch-users at lists.freeswitch.org
>> >> Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com>
>> >> Content-Type: text/plain; charset=us-ascii
>> >>
>> >> You can send the call with secure enabled and if it supports it it
>> >> will use it.
>> >>
>> >> Mike
>> >>
>> >> On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote:
>> >>
>> >>> Hello,
>> >>>
>> >>> We have a mix of phones that support RTP encryption and those that
>> >>> do not. I have to support both types in the meanwhile, and would
>> >>> like to have encryption enabled on the relevant leg, even if the
>> >>> other leg does not support it (why? one of our ATAs either must
>> >>> have it unencrypted or have it encrypted, but cannot have both).
>> >>>
>> >>> How do I find whether the destination supports encryption? I do not
>> >>> want to manage an additional table in the database...
>> >>>
>> >>
>> >>
>> >>
>> >> ------------------------------
>> >>
>> >> Message: 6
>> >> Date: Wed, 25 Nov 2009 23:25:01 +0530
>> >> From: srinivasula reddy <srinivas.ksvreddy at gmail.com>
>> >> Subject: Re: [Freeswitch-users] Bypass_media and re_invite
>> >> To: freeswitch-users at lists.freeswitch.org
>> >> Message-ID:
>> >> <f8af5740911250955x62d66f55h9584582beba76ba0 at mail.gmail.com>
>> >> Content-Type: text/plain; charset="iso-8859-1"
>> >>
>> >> HI,
>> >> thanks for your reply, my requirement is i am doing failover stuff
>> >> with
>> >> freeswitch. i dont want cut the calls when freeswitch dies, when
>> >> failover
>> >> happens mean one freeswitch dies we are going to start the second
>> >> freeswitch, i dont want close call intiated by the first
>> >> freeswtich, they
>> >> are communicating with meida(bypass media). when one endpoing try to
>> >> end the
>> >> call at that time i want to close the call for the other end also.
>> >>
>> >>
>> >> srinivas
>> >>
>> >> On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris <mike at jerris.com>
>> >> wrote:
>> >>
>> >>> FreeSWITCH will kill the calls when you shut it down, if you
>> >>> intentionally
>> >>> kill the network without shutting down FreeSWITCH the only thing
>> >>> you can do
>> >>> is enable session timers or rtp timers in the soft phones to kill
>> >>> the call
>> >>> when FreeSWITCH dies or when the call is over.
>> >>>
>> >>> Mike
>> >>>
>> >>> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote:
>> >>>
>> >>>> Hi All,
>> >>>>
>> >>>> goodmorning to all, i have a scenario, two pjsua clients are
>> >>>> connected
>> >>> with Freeswitch and they are in call and bypass_media=true. i
>> >>> close the
>> >>> Freeswitch server, still they are in call, again i started the
>> >>> Freeswitch,
>> >>> and registerd these two endpoints, now how can i end the call
>> >>> (estabilished
>> >>> by the first Freeswitch)? if i call re_invite will it estabilish
>> >>> the call
>> >>> between two endpoints?
>> >>>> any idea?
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >>> users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >>
>> >> --
>> >> Srinivasula Reddy K
>> >> -------------- next part --------------
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>> >>
>> >> ------------------------------
>> >>
>> >> Message: 7
>> >> Date: Wed, 25 Nov 2009 10:01:14 -0800
>> >> From: Stephen Crosby <stevecrozz at gmail.com>
>> >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily
>> >> response from JavaScript
>> >> To: freeswitch-users at lists.freeswitch.org
>> >> Message-ID:
>> >> <11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com>
>> >> Content-Type: text/plain; charset="utf-8"
>> >>
>> >> Surprisingly, I've found no way to access the HTTP response status
>> >> code
>> >> using mod_spidermonkey_curl. I'd love to see this feature added or
>> >> discussed
>> >> if it already exists and I'm missing it.
>> >>
>> >> --Stephen
>> >>
>> >> On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris <mike at jerris.com>
>> >> wrote:
>> >>
>> >>> In trunk there is a sofia profile setting to allow dialplan
>> >>> processing of
>> >>> 302 responses. This won't get you back into your same javascript,
>> >>> but you
>> >>> can probably do something clever from there.
>> >>>
>> >>> Mike
>> >>>
>> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote:
>> >>>
>> >>>>
>> >>>> I have considered writing JavaScript code to bridge two calls
>> >>>> together.
>> >>> However, I would like to perform custom handling of the 302 Moved
>> >>> Temporarily response. How do I handle the 302 Moved Temporarily
>> >>> response if
>> >>> I use JavaScript?
>> >>>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >>> users
>> >>> http://www.freeswitch.org
>> >>>
>> >> -------------- next part --------------
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>> >>
>> >> ------------------------------
>> >>
>> >> Message: 8
>> >> Date: Wed, 25 Nov 2009 19:04:56 +0100
>> >> From: Tihomir Culjaga <tculjaga at gmail.com>
>> >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily
>> >> response from JavaScript
>> >> To: freeswitch-users at lists.freeswitch.org
>> >> Message-ID:
>> >> <65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com>
>> >> Content-Type: text/plain; charset="iso-8859-1"
>> >>
>> >> this is how i do it from the dialplan:
>> >>
>> >>
>> >>
>> >>
>> >> <extension name="ServiceLookup">
>> >> <condition field="destination_number"
>> >> expression="^(300030)(.*)|^\+(300030)(.*)">
>> >>
>> >> <action application="set" data="bPfx=$1$3"/>
>> >> <action application="set" data="bNum=$2$4"/>
>> >>
>> >> <action inline="true" application="set"
>> >> data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/>
>> >> <action application="set"
>> >> data="caller_id_number=${cond(${intf}==true ? ${caller_id_number:
>> >> 1:32} :
>> >> ${caller_id_number})}"/>
>> >>
>> >> <action inline="true" application="set"
>> >> data="aPfx=${caller_id_number:0:6}"/>
>> >> <action inline="true" application="set"
>> >> data="aNum=${caller_id_number:6:16}"/>
>> >> <action inline="true" application="set"
>> >> data="IP_ADDR=${network_addr}:5060"/>
>> >>
>> >> <action application="lookup_service_destination" data="in $
>> >> {aNum},
>> >> in $
>> >> {aPfx},
>> >> in $
>> >> {bNum},
>> >> in $
>> >> {bPfx},
>> >> in
>> >> ${IP_ADDR},
>> >> out
>> >> redContact,
>> >> out
>> >> authResult"/>
>> >>
>> >> <action application="log" data="INFO ########################
>> >> ServiceLookup ########################\n"/>
>> >> <action application="log" data="INFO ########################
>> >> contact = '${redContact}' ##############\n"/>
>> >> <action application="log" data="INFO ########################
>> >> CallerNum = '${caller_id_number:6:16}' ##########\n"/>
>> >> <action application="log" data="INFO ########################
>> >> RADIUS auth = '${authResult}' ##########\n"/>
>> >>
>> >> <action application="execute_extension" data="doRedirect XML
>> >> public"/>
>> >> </condition>
>> >> </extension>
>> >>
>> >>
>> >> <extension name="doRedirect">
>> >> <condition field="destination_number" expression="^doRedirect$"/>
>> >> <condition field="${authResult}" expression="^0$|">
>> >> <action application="log" data="INFO ########################
>> >> RADIUS auth OK!!!' ##########\n"/>
>> >> <action application="redirect" data="${red_contact}"/>
>> >> <anti-action application="log" data="INFO
>> >> ########################
>> >> RADIUS auth NOK!! ##########\n"/>
>> >> <anti-action application="respond" data="403 Forbidden"/>
>> >> </condition>
>> >>
>> >> </extension>
>> >>
>> >>
>> >>
>> >>
>> >> On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris <mike at jerris.com>
>> >> wrote:
>> >>
>> >>> In trunk there is a sofia profile setting to allow dialplan
>> >>> processing of
>> >>> 302 responses. This won't get you back into your same javascript,
>> >>> but you
>> >>> can probably do something clever from there.
>> >>>
>> >>> Mike
>> >>>
>> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote:
>> >>>
>> >>>>
>> >>>> I have considered writing JavaScript code to bridge two calls
>> >>>> together.
>> >>> However, I would like to perform custom handling of the 302 Moved
>> >>> Temporarily response. How do I handle the 302 Moved Temporarily
>> >>> response if
>> >>> I use JavaScript?
>> >>>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >>> users
>> >>> http://www.freeswitch.org
>> >>>
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>> >>
>> >> ------------------------------
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >> users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >> End of FreeSWITCH-users Digest, Vol 41, Issue 189
>> >> *************************************************
>> >
>> > _______________________________________________
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>>
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>
>
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